Move ownership of congestion window state to rtp sender controller.

When congestion window is used, two different mechanisms can currently
update the outstanding data state in the pacer:
* OnPacketSent() withing the pacer itself, when a packet is sent
* UpdateOutstandingData(), when RtpTransportControllerSend either:
  a. Receives an OnPacketSent() callback (increase outstanding data)
  b. Receives transport feedback (decrease outstanding data)

This creates a lot of calls to UpdateOutstandingData(), more than one
per sent packet. Each requires locking and/or thread jumps. To avoid
that, this CL moves the congestion window state to
RtpTransportController send - and we only post a congested flag down
the the pacer when the state is changed.

The only benefit I can see is of the old way is we prevent sending
new packets immedately when the window is full, rather than in some
edge cases queue extra packets on the network task queue before the
congestion signal is received. That should be rare and benign.
I think this simplified logic, which is easier to read and more
performant, is a better tradeoff.

Bug: webrtc:13417
Change-Id: I326dd88db86dc0d6dc685c61920654ac024e57ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255600
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36220}
10 files changed
tree: e303ab6a03c3497ac2580b42a7e39d375e6441af
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info